Asterisk 12 One way video with Polycom RMX 2000

Asterisk Asterisk Support
1

Greetings,

I am using Asterisk version 12.5 + Webrtc2SIP and calling a Polycom RMX 2000 from SIPML5 running on the latest version of Firefox. The call completes and I am able to get two way audio and can see video from the Polycom but the bridge is recieving no video from the browser. Browser to Browser calls do complete and provide two way audio/video so it does not appear to be a firewall related issue. Asterisk+webrtc2sip is running on a VM and I am in the testing phase at the moment.

I’m wondering 1st, since Asterisk has passthrough support for the h26x video protocols, do I need the transcoding provided by webrtc2sip in order to make a call from browser to endpoint? 2nd, where is the problem in the logs attached, I reached this point about a week ago and have been stuck on getting two way video running. I just don’t see any glaring errors in the logs. Any help would be greatly appreciated.

https://dl.dropboxusercontent.com/u/71021866/Asterisk%20Debug%20Log
https://dl.dropboxusercontent.com/u/71021866/webrtc2sip.log

2

as a test I enabled the web server within asterisk and registered my web client to it. the call completes but now I am not receiving or sending video to the RMX. It looks like within the codec negotiation vp8 and opus are still being used even though they have been commented out in my setup.

v=0
o=Mozilla-SIPUA-32.0.3 24791 1 IN IP4 0.0.0.0
s=Doubango Telecom - firefox
t=0 0
a=ice-ufrag:94c8fb6b
a=ice-pwd:8d1c0c24839ede487067458f05e84847
a=fingerprint:sha-256 0C:A0:33:76:9C:8B:77:1B:92:71:7C:2A:C2:57:5F:2E:82:9F:29:05:8A:52:40:A6:29:68:27:1D:52:AD:F0:6A
m=audio 64870 UDP/TLS/RTP/SAVPF 109 0 8 101
c=IN IP4 66.210.101.146
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=setup:actpass
a=candidate:0 1 UDP 2128609535 172.23.67.209 64870 typ host
a=candidate:1 1 UDP 1692467199 66.210.101.146 64870 typ srflx raddr 172.23.67.209 rport 64870
a=candidate:0 2 UDP 2128609534 172.23.67.209 64871 typ host
a=candidate:1 2 UDP 1692467198 66.210.101.146 64871 typ srflx raddr 172.23.67.209 rport 64871
a=rtcp-mux
m=video 64872 UDP/TLS/RTP/SAVPF 120
c=IN IP4 66.210.101.146
a=rtpmap:120 VP8/90000
a=sendrecv
a=rtcp-fb:120 nack
a=rtcp-fb:120 nack pli
a=rtcp-fb:120 ccm fir
a=setup:actpass
a=candidate:0 1 UDP 2128609535 172.23.67.209 64872 typ host
a=candidate:1 1 UDP 1692467199 66.210.101.146 64872 typ srflx raddr 172.23.67.209 rport 64872
a=candidate:0 2 UDP 2128609534 172.23.67.209 64873 typ host
a=candidate:1 2 UDP 1692467198 66.210.101.146 64873 typ srflx raddr 172.23.67.209 rport 64873
a=rtcp-mux

sip.conf

[1060]
type=friend
defaultuser=1060
host=dynamic
secret=1060
;encryption=yes
avpf=yes
icesupport=yes
context=public
directmedia=no
videosupport=yes
disallow=all
;allow=opus
;allow=g729
allow=g726
allow=speex
allow=g722
allow=ulaw
allow=alaw
;allow=vp8
allow=h264
allow=h263p
allow=h263
allow=h261
transport=udp,ws
force_avp=yes
dtlsenable=yes
dtlsverify=no
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlsprivatekey=/etc/asterisk/keys/asterisk.pem
dtlssetup=actpass

3

You’ve only provided the SDP from the peer, not from Asterisk. There is no evidence that Asterisk is offering to use VP8.

4

David, If that’s not included in the asterisk debug log where would I find it?

thanks for the response

5

There should be SDP in both directions.

6

this should include both the offer and answer:

SEND: INVITE sip:14172583@mydomain.tld SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKD17JUm0q8t8QAuFkHUNJIKfdOL6c0fcF;rport
From: "1060"sip:1060@mydomain.tld;tag=xm7QiFtHWs8E561qgANT
To: sip:14172583@mydomain.tld
Contact: "1060"sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws;impi=1060;ha1=ef80cafc77d8e41a49bf4d312bc51e1d;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c1d00f9b-3ec4-6ff0-d991-3122ff875858
CSeq: 35491 INVITE
Content-Type: application/sdp
Content-Length: 1392
Route: sip:mydomain.tld:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

v=0
o=Mozilla-SIPUA-32.0.3 180 1 IN IP4 0.0.0.0
s=Doubango Telecom - firefox
t=0 0
a=ice-ufrag:8fe5500d
a=ice-pwd:1aeffea68fb418d899e7aef584e33aef
a=fingerprint:sha-256 B8:0D:09:27:C6:DB:E6:01:C5:66:3A:91:E5:7B:09:F8:1E:A5:DC:7D:E9:5C:3D:41:7B:3E:79:DA:71:B0:BF:CB
m=audio 52058 UDP/TLS/RTP/SAVPF 109 0 8 101
c=IN IP4 66.210.101.146
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=setup:actpass
a=candidate:0 1 UDP 2128609535 172.23.67.209 52058 typ host
a=candidate:1 1 UDP 1692467199 66.210.101.146 52058 typ srflx raddr 172.23.67.209 rport 52058
a=candidate:0 2 UDP 2128609534 172.23.67.209 52059 typ host
a=candidate:1 2 UDP 1692467198 66.210.101.146 52059 typ srflx raddr 172.23.67.209 rport 52059
a=rtcp-mux
m=video 52060 UDP/TLS/RTP/SAVPF 120
c=IN IP4 66.210.101.146
a=rtpmap:120 VP8/90000
a=sendrecv
a=rtcp-fb:120 nack
a=rtcp-fb:120 nack pli
a=rtcp-fb:120 ccm fir
a=setup:actpass
a=candidate:0 1 UDP 2128609535 172.23.67.209 52060 typ host
a=candidate:1 1 UDP 1692467199 66.210.101.146 52060 typ srflx raddr 172.23.67.209 rport 52060
a=candidate:0 2 UDP 2128609534 172.23.67.209 52061 typ host
a=candidate:1 2 UDP 1692467198 66.210.101.146 52061 typ srflx raddr 172.23.67.209 rport 52061
a=rtcp-mux

tsk_uti…svn=224 (line 116)
onIceCandidate = undefined
tsk_uti…svn=224 (line 116)
ICE GATHERING COMPLETED!
tsk_uti…svn=224 (line 116)
onIceGatheringCompleted
tsk_uti…svn=224 (line 116)
onIceGatheringCompleted but no local sdp request is pending
tsk_uti…svn=224 (line 122)
__tsip_transport_ws_onmessage
tsk_uti…svn=224 (line 116)

recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=10.0.2.2;branch=z9hG4bKD17JUm0q8t8QAuFkHUNJIKfdOL6c0fcF
From: "1060"sip:1060@mydomain.tld;tag=xm7QiFtHWs8E561qgANT
To: sip:14172583@mydomain.tld;tag=as5b801068
Call-ID: c1d00f9b-3ec4-6ff0-d991-3122ff875858
CSeq: 35491 INVITE
Content-Length: 0
Server: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm=“mydomain.tld”,nonce=“142c274d”,stale=FALSE,algorithm=MD5

tsk_uti…svn=224 (line 116)

SEND: ACK sip:14172583@mydomain.tld SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKD17JUm0q8t8QAuFkHUNJIKfdOL6c0fcF;rport
From: "1060"sip:1060@mydomain.tld;tag=xm7QiFtHWs8E561qgANT
To: sip:14172583@mydomain.tld;tag=as5b801068
Call-ID: c1d00f9b-3ec4-6ff0-d991-3122ff875858
CSeq: 35491 ACK
Content-Length: 0
Route: sip:mydomain.tld:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70

tsk_uti…svn=224 (line 116)
State machine: x0000_Any_2_Any_X_i401_407_INVITE
tsk_uti…svn=224 (line 116)

SEND: INVITE sip:14172583@mydomain.tld SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKZMPdr9LlEBo1X8Kv1vVuQx52bUu2NozZ;rport
From: "1060"sip:1060@mydomain.tld;tag=xm7QiFtHWs8E561qgANT
To: sip:14172583@mydomain.tld
Contact: "1060"sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws;impi=1060;ha1=ef80cafc77d8e41a49bf4d312bc51e1d;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c1d00f9b-3ec4-6ff0-d991-3122ff875858
CSeq: 35492 INVITE
Content-Type: application/sdp
Content-Length: 1392
Route: sip:mydomain.tld:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“mydomain.tld”,nonce=“142c274d”,uri=“sip:14172583@mydomain.tld”,response=“b0707622d9f3853bce52dffd5099af84”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

v=0
o=Mozilla-SIPUA-32.0.3 180 1 IN IP4 0.0.0.0
s=Doubango Telecom - firefox
t=0 0
a=ice-ufrag:8fe5500d
a=ice-pwd:1aeffea68fb418d899e7aef584e33aef
a=fingerprint:sha-256 B8:0D:09:27:C6:DB:E6:01:C5:66:3A:91:E5:7B:09:F8:1E:A5:DC:7D:E9:5C:3D:41:7B:3E:79:DA:71:B0:BF:CB
m=audio 52058 UDP/TLS/RTP/SAVPF 109 0 8 101
c=IN IP4 66.210.101.146
a=rtpmap:109 opus/48000/2
a=ptime:20
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=setup:actpass
a=candidate:0 1 UDP 2128609535 172.23.67.209 52058 typ host
a=candidate:1 1 UDP 1692467199 66.210.101.146 52058 typ srflx raddr 172.23.67.209 rport 52058
a=candidate:0 2 UDP 2128609534 172.23.67.209 52059 typ host
a=candidate:1 2 UDP 1692467198 66.210.101.146 52059 typ srflx raddr 172.23.67.209 rport 52059
a=rtcp-mux
m=video 52060 UDP/TLS/RTP/SAVPF 120
c=IN IP4 66.210.101.146
a=rtpmap:120 VP8/90000
a=sendrecv
a=rtcp-fb:120 nack
a=rtcp-fb:120 nack pli
a=rtcp-fb:120 ccm fir
a=setup:actpass
a=candidate:0 1 UDP 2128609535 172.23.67.209 52060 typ host
a=candidate:1 1 UDP 1692467199 66.210.101.146 52060 typ srflx raddr 172.23.67.209 rport 52060
a=candidate:0 2 UDP 2128609534 172.23.67.209 52061 typ host
a=candidate:1 2 UDP 1692467198 66.210.101.146 52061 typ srflx raddr 172.23.67.209 rport 52061
a=rtcp-mux

tsk_uti…svn=224 (line 116)
__tsip_transport_ws_onmessage
tsk_uti…svn=224 (line 116)

recv=SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=10.0.2.2;branch=z9hG4bKZMPdr9LlEBo1X8Kv1vVuQx52bUu2NozZ
From: "1060"sip:1060@mydomain.tld;tag=xm7QiFtHWs8E561qgANT
To: sip:14172583@mydomain.tld
Contact: sip:14172583@10.0.2.15:5060;transport=WS
Call-ID: c1d00f9b-3ec4-6ff0-d991-3122ff875858
CSeq: 35492 INVITE
Content-Length: 0
Server: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

tsk_uti…svn=224 (line 116)
State machine: x0000_Any_2_Any_X_i1xx
tsk_uti…svn=224 (line 116)
==session event = i_ao_request
tsk_uti…svn=224 (line 116)
__tsip_transport_ws_onmessage
tsk_uti…svn=224 (line 116)

recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=10.0.2.2;branch=z9hG4bKZMPdr9LlEBo1X8Kv1vVuQx52bUu2NozZ
From: "1060"sip:1060@mydomain.tld;tag=xm7QiFtHWs8E561qgANT
To: sip:14172583@mydomain.tld;tag=as0082401e
Contact: sip:14172583@10.0.2.15:5060;transport=WS
Call-ID: c1d00f9b-3ec4-6ff0-d991-3122ff875858
CSeq: 35492 INVITE
Content-Length: 0
Server: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

tsk_uti…svn=224 (line 116)
State machine: x0000_Any_2_Any_X_i1xx
tsk_uti…svn=224 (line 116)
==session event = i_ao_request
tsk_uti…svn=224 (line 116)
__tsip_transport_ws_onmessage
tsk_uti…svn=224 (line 116)

recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=10.0.2.2;branch=z9hG4bKZMPdr9LlEBo1X8Kv1vVuQx52bUu2NozZ
From: "1060"sip:1060@mydomain.tld;tag=xm7QiFtHWs8E561qgANT
To: sip:14172583@mydomain.tld;tag=as0082401e
Contact: sip:14172583@10.0.2.15:5060;transport=WS
Call-ID: c1d00f9b-3ec4-6ff0-d991-3122ff875858
CSeq: 35492 INVITE
Content-Type: application/sdp
Content-Length: 815
Server: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

v=0
o=root 189627515 189627515 IN IP4 10.0.2.15
s=Asterisk PBX 12.5.1
c=IN IP4 10.0.2.15
t=0 0
m=audio 10082 RTP/SAVPF 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=ice-ufrag:13d9bbd0194a839a1d57b5cf3b1d54e0
a=ice-pwd:484de2704618f21c3a77702f52d566aa
a=candidate:Ha00020f 1 UDP 2130706431 10.0.2.15 10082 typ host
a=candidate:S42d26592 1 UDP 1694498815 66.210.101.146 10082 typ srflx
a=candidate:Ha00020f 2 UDP 2130706430 10.0.2.15 10083 typ host
a=candidate:S42d26592 2 UDP 1694498814 66.210.101.146 10084 typ srflx
a=connection:new
a=setup:active
a=fingerprint:SHA-256 27:2B:A6:AF:74:97:B5:91:98:B1:E9:6A:63:C0:DD:FB:BB:20:36:5E:4F:33:18:D2:FB:76:5C:A2:08:44:75:70
a=sendrecv
m=video 0 UDP/TLS/RTP/SAVPF 120

tsk_uti…svn=224 (line 116)
State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE
tsk_uti…svn=224 (line 116)

setRemoteDescription(answer)
v=0
o=root 189627515 189627515 IN IP4 10.0.2.15
s=Asterisk PBX 12.5.1
c=IN IP4 10.0.2.15
t=0 0
m=audio 10082 RTP/SAVPF 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=ice-ufrag:13d9bbd0194a839a1d57b5cf3b1d54e0
a=ice-pwd:484de2704618f21c3a77702f52d566aa
a=candidate:Ha00020f 1 UDP 2130706431 10.0.2.15 10082 typ host
a=candidate:S42d26592 1 UDP 1694498815 66.210.101.146 10082 typ srflx
a=candidate:Ha00020f 2 UDP 2130706430 10.0.2.15 10083 typ host
a=candidate:S42d26592 2 UDP 1694498814 66.210.101.146 10084 typ srflx
a=connection:new
a=setup:active
a=fingerprint:SHA-256 27:2B:A6:AF:74:97:B5:91:98:B1:E9:6A:63:C0:DD:FB:BB:20:36:5E:4F:33:18:D2:FB:76:5C:A2:08:44:75:70
a=sendrecv
m=video 0 RTP/SAVPF 120

tsk_uti…svn=224 (line 116)

SEND: ACK sip:14172583@10.0.2.15:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK0smHEkvEJtyygyhMiNKW;rport
From: "1060"sip:1060@mydomain.tld;tag=xm7QiFtHWs8E561qgANT
To: sip:14172583@mydomain.tld;tag=as0082401e
Contact: "1060"sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: c1d00f9b-3ec4-6ff0-d991-3122ff875858
CSeq: 35492 ACK
Content-Length: 0
Route: sip:mydomain.tld:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“mydomain.tld”,nonce=“142c274d”,uri=“sip:14172583@10.0.2.15:5060;transport=WS”,response=“2aefacce9552b914858119afd9903242”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

tsk_uti…svn=224 (line 116)
==session event = m_early_media
tsk_uti…svn=224 (line 116)
==session event = connected
tsk_uti…svn=224 (line 116)
__on_add_stream
tsk_uti…svn=224 (line 116)
onSetRemoteDescriptionSuccess
tsk_uti…svn=224 (line 116)
==session event = m_stream_audio_remote_added
tsk_uti…svn=224 (line 116)
__tsip_transport_ws_onmessage
tsk_uti…svn=224 (line 116)

recv=INFO sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.0.2.15:5060;branch=z9hG4bK6335d46b
From: sip:14172583@mydomain.tld;tag=as0082401e
To: "1060"sip:1060@mydomain.tld;tag=xm7QiFtHWs8E561qgANT
Contact: sip:14172583@10.0.2.15:5060;transport=WS
Call-ID: c1d00f9b-3ec4-6ff0-d991-3122ff875858
CSeq: 102 INFO
Content-Type: application/media_control+xml
Content-Length: 205
Max-Forwards: 70
User-Agent: Asterisk PBX

<?xml version="1.0" encoding="utf-8" ?>

<media_control>
<vc_primitive>
<to_encoder>
<picture_fast_update>
</picture_fast_update>
</to_encoder>
</vc_primitive>
</media_control>

tsk_uti…svn=224 (line 116)
State machine: x0000_Any_2_Any_X_iINFO
tsk_uti…svn=224 (line 116)

SEND: SIP/2.0 200 OK
Via: SIP/2.0/WS 10.0.2.15:5060;branch=z9hG4bK6335d46b
From: sip:14172583@mydomain.tld;tag=as0082401e
To: "1060"sip:1060@mydomain.tld;tag=xm7QiFtHWs8E561qgANT
Contact: sip:1060@df7jal23ls0d.invalid;transport=ws
Call-ID: c1d00f9b-3ec4-6ff0-d991-3122ff875858
CSeq: 102 INFO
Content-Length: 0

tsk_uti…svn=224 (line 116)
==session event = i_info
tsk_uti…svn=224 (line 116)
__tsip_transport_ws_onmessage
tsk_uti…svn=224 (line 116)

recv=INFO sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.0.2.15:5060;branch=z9hG4bK5ec07234
From: sip:14172583@mydomain.tld;tag=as0082401e
To: "1060"sip:1060@mydomain.tld;tag=xm7QiFtHWs8E561qgANT
Contact: sip:14172583@10.0.2.15:5060;transport=WS
Call-ID: c1d00f9b-3ec4-6ff0-d991-3122ff875858
CSeq: 103 INFO
Content-Type: application/media_control+xml
Content-Length: 205
Max-Forwards: 70
User-Agent: Asterisk PBX

<?xml version="1.0" encoding="utf-8" ?>

<media_control>
<vc_primitive>
<to_encoder>
<picture_fast_update>
</picture_fast_update>
</to_encoder>
</vc_primitive>
</media_control>

tsk_uti…svn=224 (line 116)
State machine: x0000_Any_2_Any_X_iINFO
tsk_uti…svn=224 (line 116)

SEND: SIP/2.0 200 OK
Via: SIP/2.0/WS 10.0.2.15:5060;branch=z9hG4bK5ec07234
From: sip:14172583@mydomain.tld;tag=as0082401e
To: "1060"sip:1060@mydomain.tld;tag=xm7QiFtHWs8E561qgANT
Contact: sip:1060@df7jal23ls0d.invalid;transport=ws
Call-ID: c1d00f9b-3ec4-6ff0-d991-3122ff875858
CSeq: 103 INFO
Content-Length: 0

tsk_uti…svn=224 (line 116)
==session event = i_info
tsk_uti…svn=224 (line 116)
__tsip_transport_ws_onmessage
tsk_uti…svn=224 (line 116)

recv=INFO sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.0.2.15:5060;branch=z9hG4bK5ee8b793
From: sip:14172583@mydomain.tld;tag=as0082401e
To: "1060"sip:1060@mydomain.tld;tag=xm7QiFtHWs8E561qgANT
Contact: sip:14172583@10.0.2.15:5060;transport=WS
Call-ID: c1d00f9b-3ec4-6ff0-d991-3122ff875858
CSeq: 104 INFO
Content-Type: application/media_control+xml
Content-Length: 205
Max-Forwards: 70
User-Agent: Asterisk PBX

<?xml version="1.0" encoding="utf-8" ?>

<media_control>
<vc_primitive>
<to_encoder>
<picture_fast_update>
</picture_fast_update>
</to_encoder>
</vc_primitive>
</media_control>

tsk_uti…svn=224 (line 116)
State machine: x0000_Any_2_Any_X_iINFO
tsk_uti…svn=224 (line 116)

SEND: SIP/2.0 200 OK
Via: SIP/2.0/WS 10.0.2.15:5060;branch=z9hG4bK5ee8b793
From: sip:14172583@mydomain.tld;tag=as0082401e
To: "1060"sip:1060@mydomain.tld;tag=xm7QiFtHWs8E561qgANT
Contact: sip:1060@df7jal23ls0d.invalid;transport=ws
Call-ID: c1d00f9b-3ec4-6ff0-d991-3122ff875858
CSeq: 104 INFO
Content-Length: 0

tsk_uti…svn=224 (line 116)
==session event = i_info
tsk_uti…svn=224 (line 116)
__tsip_transport_ws_onmessage
tsk_uti…svn=224 (line 116)

recv=INFO sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.0.2.15:5060;branch=z9hG4bK33d16dfe
From: sip:14172583@mydomain.tld;tag=as0082401e
To: "1060"sip:1060@mydomain.tld;tag=xm7QiFtHWs8E561qgANT
Contact: sip:14172583@10.0.2.15:5060;transport=WS
Call-ID: c1d00f9b-3ec4-6ff0-d991-3122ff875858
CSeq: 105 INFO
Content-Type: application/media_control+xml
Content-Length: 205
Max-Forwards: 70
User-Agent: Asterisk PBX

<?xml version="1.0" encoding="utf-8" ?>

<media_control>
<vc_primitive>
<to_encoder>
<picture_fast_update>
</picture_fast_update>
</to_encoder>
</vc_primitive>
</media_control>

tsk_uti…svn=224 (line 116)
State machine: x0000_Any_2_Any_X_iINFO
tsk_uti…svn=224 (line 116)

SEND: SIP/2.0 200 OK
Via: SIP/2.0/WS 10.0.2.15:5060;branch=z9hG4bK33d16dfe
From: sip:14172583@mydomain.tld;tag=as0082401e
To: "1060"sip:1060@mydomain.tld;tag=xm7QiFtHWs8E561qgANT
Contact: sip:1060@df7jal23ls0d.invalid;transport=ws
Call-ID: c1d00f9b-3ec4-6ff0-d991-3122ff875858
CSeq: 105 INFO
Content-Length: 0

tsk_uti…svn=224 (line 116)
==session event = i_info
tsk_uti…svn=224 (line 116)
__tsip_transport_ws_onmessage
tsk_uti…svn=224 (line 116)

recv=INFO sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.0.2.15:5060;branch=z9hG4bK4083dc44
From: sip:14172583@mydomain.tld;tag=as0082401e
To: "1060"sip:1060@mydomain.tld;tag=xm7QiFtHWs8E561qgANT
Contact: sip:14172583@10.0.2.15:5060;transport=WS
Call-ID: c1d00f9b-3ec4-6ff0-d991-3122ff875858
CSeq: 106 INFO
Content-Type: application/media_control+xml
Content-Length: 205
Max-Forwards: 70
User-Agent: Asterisk PBX

<?xml version="1.0" encoding="utf-8" ?>

<media_control>
<vc_primitive>
<to_encoder>
<picture_fast_update>
</picture_fast_update>
</to_encoder>
</vc_primitive>
</media_control>

tsk_uti…svn=224 (line 116)
State machine: x0000_Any_2_Any_X_iINFO
tsk_uti…svn=224 (line 116)

SEND: SIP/2.0 200 OK
Via: SIP/2.0/WS 10.0.2.15:5060;branch=z9hG4bK4083dc44
From: sip:14172583@mydomain.tld;tag=as0082401e
To: "1060"sip:1060@mydomain.tld;tag=xm7QiFtHWs8E561qgANT
Contact: sip:1060@df7jal23ls0d.invalid;transport=ws
Call-ID: c1d00f9b-3ec4-6ff0-d991-3122ff875858
CSeq: 106 INFO
Content-Length: 0

tsk_uti…svn=224 (line 116)
==session event = i_info
tsk_uti…svn=224 (line 116)
__tsip_transport_ws_onmessage
tsk_uti…svn=224 (line 116)

recv=INFO sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.0.2.15:5060;branch=z9hG4bK33f852a5
From: sip:14172583@mydomain.tld;tag=as0082401e
To: "1060"sip:1060@mydomain.tld;tag=xm7QiFtHWs8E561qgANT
Contact: sip:14172583@10.0.2.15:5060;transport=WS
Call-ID: c1d00f9b-3ec4-6ff0-d991-3122ff875858
CSeq: 107 INFO
Content-Type: application/media_control+xml
Content-Length: 205
Max-Forwards: 70
User-Agent: Asterisk PBX

<?xml version="1.0" encoding="utf-8" ?>

<media_control>
<vc_primitive>
<to_encoder>
<picture_fast_update>
</picture_fast_update>
</to_encoder>
</vc_primitive>
</media_control>

tsk_uti…svn=224 (line 116)
State machine: x0000_Any_2_Any_X_iINFO
tsk_uti…svn=224 (line 116)

SEND: SIP/2.0 200 OK
Via: SIP/2.0/WS 10.0.2.15:5060;branch=z9hG4bK33f852a5
From: sip:14172583@mydomain.tld;tag=as0082401e
To: "1060"sip:1060@mydomain.tld;tag=xm7QiFtHWs8E561qgANT
Contact: sip:1060@df7jal23ls0d.invalid;transport=ws
Call-ID: c1d00f9b-3ec4-6ff0-d991-3122ff875858
CSeq: 107 INFO
Content-Length: 0

tsk_uti…svn=224 (line 116)
==session event = i_info
tsk_uti…svn=224 (line 116)
==session event = m_stream_video_local_added
tsk_uti…svn=224 (line 116)
==session event = m_stream_audio_local_added
tsk_uti…svn=224 (line 116)
==session event = m_stream_audio_remote_added
tsk_uti…svn=224 (line 116)
State machine: x0000_Any_2_Trying_X_oBYE
tsk_uti…svn=224 (line 116)

SEND: BYE sip:14172583@10.0.2.15:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKXxhAntCTPzuAGEKVWQToFT97ETgBCk5l;rport
From: "1060"sip:1060@mydomain.tld;tag=xm7QiFtHWs8E561qgANT
To: sip:14172583@mydomain.tld;tag=as0082401e
Call-ID: c1d00f9b-3ec4-6ff0-d991-3122ff875858
CSeq: 35493 BYE
Content-Length: 0
Route: sip:mydomain.tld:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70
Accept-Contact: *;+g.oma.sip-im
Accept-Contact: *;+sip.ice
Accept-Contact: *;language="en,fr"
Authorization: Digest username=“1060”,realm=“mydomain.tld”,nonce=“142c274d”,uri=“sip:14172583@10.0.2.15:5060;transport=WS”,response=“367f9f7f995dd3aa06f178eca0deb2a5”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

7

from what I can see in the last SDP is that my client is offering VP8 even though it’s been disabled in the user profile, and it only offers VP8 although the h26x codecs are all enabled. The answer from the RMX does not appear to include any video codec since the Polycom systems don’t support VP8. The Polycom still sends H263 and the web client can see it.

8

It starts with Mozilla offering VP8 only. Asterisk rejects this by setting the port number to zero. Result no video enabled and Asterisk is correctly honouring the lack of an allow for VP8.

9

is there a way to change this behaviour? My sip.conf doesn’t have vp8 as an allowed codec and I believe Firefox has support for at least h261, which is enabled, why isn’t that codec a part of the offer.

10

Firefox issue.

11

got it thanks, that explains a lot. I guess i’ll go back to working in chrome then since it’s the only other browser with good RTC support.

12

tested again with Chrome Canary (40.0.2180.0) and Chrome 37 with the same result, VP8 is the only offered codec therefor no video flows:

ICE GATHERING COMPLETED!
tsk_utils.js?svn=224:116 onIceGatheringCompleted
tsk_utils.js?svn=224:116 SEND: INVITE sip:14172583@mydomain.tld SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKtTdJbKvkxndkJInyBR95T2yKcUHuXii4;rport
From: "1060"sip:1060@mydomain.tld;tag=U5d8C1sd4CRTkQQHTiCh
To: sip:14172583@mydomain.tld
Contact: "1060"sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 2b9ea5b5-41e9-30ac-fbcc-3f83e9f2ef1e
CSeq: 2263 INVITE
Content-Type: application/sdp
Content-Length: 3662
Route: sip:mydomain.tld:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

v=0
o=- 8147320829078889000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS lYBFiwCUN8pEhJ3YMkSFRZvVeTOEvYOnhfoz
m=audio 52520 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 66.210.101.146
a=rtcp:52520 IN IP4 66.210.101.146
a=candidate:1480854522 1 udp 2122260223 172.23.67.209 52520 typ host generation 0
a=candidate:1480854522 2 udp 2122260223 172.23.67.209 52520 typ host generation 0
a=candidate:3641445710 1 udp 1686052607 66.210.101.146 52520 typ srflx raddr 172.23.67.209 rport 52520 generation 0
a=candidate:3641445710 2 udp 1686052607 66.210.101.146 52520 typ srflx raddr 172.23.67.209 rport 52520 generation 0
a=candidate:382203658 1 tcp 1518280447 172.23.67.209 0 typ host tcptype active generation 0
a=candidate:382203658 2 tcp 1518280447 172.23.67.209 0 typ host tcptype active generation 0
a=ice-ufrag:DKzWXuXldGKfxdOP
a=ice-pwd:yoeCl/NPJt27Y63DjrQ7/YOF
a=ice-options:google-ice
a=fingerprint:sha-256 A6:2E:95:0C:20:B2:E7:03:14:D4:91:A9:18:4C:48:09:7A:0F:C5:47:1F:55:58:BA:7E:44:0E:3F:A2:70:24:8E
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 webrtc.org/experiments/rtp-h … -send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:1599748318 cname:Tl7uG2ihegOYlauy
a=ssrc:1599748318 msid:lYBFiwCUN8pEhJ3YMkSFRZvVeTOEvYOnhfoz f3954b8d-6d24-4218-a93c-2b42a46bb0fe
a=ssrc:1599748318 mslabel:lYBFiwCUN8pEhJ3YMkSFRZvVeTOEvYOnhfoz
a=ssrc:1599748318 label:f3954b8d-6d24-4218-a93c-2b42a46bb0fe
m=video 52520 UDP/TLS/RTP/SAVPF 100 116 117 96
c=IN IP4 66.210.101.146
a=rtcp:52520 IN IP4 66.210.101.146
a=candidate:1480854522 1 udp 2122260223 172.23.67.209 52520 typ host generation 0
a=candidate:1480854522 2 udp 2122260223 172.23.67.209 52520 typ host generation 0
a=candidate:3641445710 1 udp 1686052607 66.210.101.146 52520 typ srflx raddr 172.23.67.209 rport 52520 generation 0
a=candidate:3641445710 2 udp 1686052607 66.210.101.146 52520 typ srflx raddr 172.23.67.209 rport 52520 generation 0
a=candidate:382203658 1 tcp 1518280447 172.23.67.209 0 typ host tcptype active generation 0
a=candidate:382203658 2 tcp 1518280447 172.23.67.209 0 typ host tcptype active generation 0
a=ice-ufrag:DKzWXuXldGKfxdOP
a=ice-pwd:yoeCl/NPJt27Y63DjrQ7/YOF
a=ice-options:google-ice
a=fingerprint:sha-256 A6:2E:95:0C:20:B2:E7:03:14:D4:91:A9:18:4C:48:09:7A:0F:C5:47:1F:55:58:BA:7E:44:0E:3F:A2:70:24:8E
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 webrtc.org/experiments/rtp-h … -send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:96 rtx/90000
a=fmtp:96 apt=100
a=ssrc-group:FID 1821200420 2301671825
a=ssrc:1821200420 cname:Tl7uG2ihegOYlauy
a=ssrc:1821200420 msid:lYBFiwCUN8pEhJ3YMkSFRZvVeTOEvYOnhfoz 73c70abf-6435-4dbb-b9a9-5436bb2847a0
a=ssrc:1821200420 mslabel:lYBFiwCUN8pEhJ3YMkSFRZvVeTOEvYOnhfoz
a=ssrc:1821200420 label:73c70abf-6435-4dbb-b9a9-5436bb2847a0
a=ssrc:2301671825 cname:Tl7uG2ihegOYlauy
a=ssrc:2301671825 msid:lYBFiwCUN8pEhJ3YMkSFRZvVeTOEvYOnhfoz 73c70abf-6435-4dbb-b9a9-5436bb2847a0
a=ssrc:2301671825 mslabel:lYBFiwCUN8pEhJ3YMkSFRZvVeTOEvYOnhfoz
a=ssrc:2301671825 label:73c70abf-6435-4dbb-b9a9-5436bb2847a0

tsk_utils.js?svn=224:116 __tsip_transport_ws_onmessage
tsk_utils.js?svn=224:116 recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.2.15:8088;rport=8088;received=10.0.2.15;branch=z9hG4bKtTdJbKvkxndkJInyBR95T2yKcUHuXii4
From: "1060"sip:1060@mydomain.tld;tag=U5d8C1sd4CRTkQQHTiCh
To: sip:14172583@mydomain.tld;tag=as23f5b0df
Call-ID: 2b9ea5b5-41e9-30ac-fbcc-3f83e9f2ef1e
CSeq: 2263 INVITE
Content-Length: 0
Via: SIP/2.0/TCP 10.0.2.2:49497;rport;branch=z9hG4bKtTdJbKvkxndkJInyBR95T2yKcUHuXii4;ws-hacked=WS
Server: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer
WWW-Authenticate: Digest realm=“mydomain.tld”,nonce=“5c3eb758”,stale=FALSE,algorithm=MD5

tsk_utils.js?svn=224:116 SEND: ACK sip:14172583@mydomain.tld SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKtTdJbKvkxndkJInyBR95T2yKcUHuXii4;rport
From: "1060"sip:1060@mydomain.tld;tag=U5d8C1sd4CRTkQQHTiCh
To: sip:14172583@mydomain.tld;tag=as23f5b0df
Call-ID: 2b9ea5b5-41e9-30ac-fbcc-3f83e9f2ef1e
CSeq: 2263 ACK
Content-Length: 0
Route: sip:mydomain.tld:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70

tsk_utils.js?svn=224:116 State machine: x0000_Any_2_Any_X_i401_407_INVITE
tsk_utils.js?svn=224:116 SEND: INVITE sip:14172583@mydomain.tld SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKcpNgLRRAW87qugiSncmycKIPypHCh5nH;rport
From: "1060"sip:1060@mydomain.tld;tag=U5d8C1sd4CRTkQQHTiCh
To: sip:14172583@mydomain.tld
Contact: "1060"sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=ws;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 2b9ea5b5-41e9-30ac-fbcc-3f83e9f2ef1e
CSeq: 2264 INVITE
Content-Type: application/sdp
Content-Length: 3662
Route: sip:mydomain.tld:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70
Authorization: Digest username=“1060”,realm=“mydomain.tld”,nonce=“5c3eb758”,uri=“sip:14172583@mydomain.tld”,response=“f0b66234b0aeea9557b97f734bda77fd”,algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2014.04.18
Organization: Doubango Telecom

v=0
o=- 8147320829078889000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio video
a=msid-semantic: WMS lYBFiwCUN8pEhJ3YMkSFRZvVeTOEvYOnhfoz
m=audio 52520 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 66.210.101.146
a=rtcp:52520 IN IP4 66.210.101.146
a=candidate:1480854522 1 udp 2122260223 172.23.67.209 52520 typ host generation 0
a=candidate:1480854522 2 udp 2122260223 172.23.67.209 52520 typ host generation 0
a=candidate:3641445710 1 udp 1686052607 66.210.101.146 52520 typ srflx raddr 172.23.67.209 rport 52520 generation 0
a=candidate:3641445710 2 udp 1686052607 66.210.101.146 52520 typ srflx raddr 172.23.67.209 rport 52520 generation 0
a=candidate:382203658 1 tcp 1518280447 172.23.67.209 0 typ host tcptype active generation 0
a=candidate:382203658 2 tcp 1518280447 172.23.67.209 0 typ host tcptype active generation 0
a=ice-ufrag:DKzWXuXldGKfxdOP
a=ice-pwd:yoeCl/NPJt27Y63DjrQ7/YOF
a=ice-options:google-ice
a=fingerprint:sha-256 A6:2E:95:0C:20:B2:E7:03:14:D4:91:A9:18:4C:48:09:7A:0F:C5:47:1F:55:58:BA:7E:44:0E:3F:A2:70:24:8E
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 webrtc.org/experiments/rtp-h … -send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:1599748318 cname:Tl7uG2ihegOYlauy
a=ssrc:1599748318 msid:lYBFiwCUN8pEhJ3YMkSFRZvVeTOEvYOnhfoz f3954b8d-6d24-4218-a93c-2b42a46bb0fe
a=ssrc:1599748318 mslabel:lYBFiwCUN8pEhJ3YMkSFRZvVeTOEvYOnhfoz
a=ssrc:1599748318 label:f3954b8d-6d24-4218-a93c-2b42a46bb0fe
m=video 52520 UDP/TLS/RTP/SAVPF 100 116 117 96
c=IN IP4 66.210.101.146
a=rtcp:52520 IN IP4 66.210.101.146
a=candidate:1480854522 1 udp 2122260223 172.23.67.209 52520 typ host generation 0
a=candidate:1480854522 2 udp 2122260223 172.23.67.209 52520 typ host generation 0
a=candidate:3641445710 1 udp 1686052607 66.210.101.146 52520 typ srflx raddr 172.23.67.209 rport 52520 generation 0
a=candidate:3641445710 2 udp 1686052607 66.210.101.146 52520 typ srflx raddr 172.23.67.209 rport 52520 generation 0
a=candidate:382203658 1 tcp 1518280447 172.23.67.209 0 typ host tcptype active generation 0
a=candidate:382203658 2 tcp 1518280447 172.23.67.209 0 typ host tcptype active generation 0
a=ice-ufrag:DKzWXuXldGKfxdOP
a=ice-pwd:yoeCl/NPJt27Y63DjrQ7/YOF
a=ice-options:google-ice
a=fingerprint:sha-256 A6:2E:95:0C:20:B2:E7:03:14:D4:91:A9:18:4C:48:09:7A:0F:C5:47:1F:55:58:BA:7E:44:0E:3F:A2:70:24:8E
a=setup:actpass
a=mid:video
a=extmap:2 urn:ietf:params:rtp-hdrext:toffset
a=extmap:3 webrtc.org/experiments/rtp-h … -send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:100 VP8/90000
a=rtcp-fb:100 ccm fir
a=rtcp-fb:100 nack
a=rtcp-fb:100 nack pli
a=rtcp-fb:100 goog-remb
a=rtpmap:116 red/90000
a=rtpmap:117 ulpfec/90000
a=rtpmap:96 rtx/90000
a=fmtp:96 apt=100
a=ssrc-group:FID 1821200420 2301671825
a=ssrc:1821200420 cname:Tl7uG2ihegOYlauy
a=ssrc:1821200420 msid:lYBFiwCUN8pEhJ3YMkSFRZvVeTOEvYOnhfoz 73c70abf-6435-4dbb-b9a9-5436bb2847a0
a=ssrc:1821200420 mslabel:lYBFiwCUN8pEhJ3YMkSFRZvVeTOEvYOnhfoz
a=ssrc:1821200420 label:73c70abf-6435-4dbb-b9a9-5436bb2847a0
a=ssrc:2301671825 cname:Tl7uG2ihegOYlauy
a=ssrc:2301671825 msid:lYBFiwCUN8pEhJ3YMkSFRZvVeTOEvYOnhfoz 73c70abf-6435-4dbb-b9a9-5436bb2847a0
a=ssrc:2301671825 mslabel:lYBFiwCUN8pEhJ3YMkSFRZvVeTOEvYOnhfoz
a=ssrc:2301671825 label:73c70abf-6435-4dbb-b9a9-5436bb2847a0

tsk_utils.js?svn=224:116 __tsip_transport_ws_onmessage
tsk_utils.js?svn=224:116 recv=SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 10.0.2.15:8088;rport=8088;received=10.0.2.15;branch=z9hG4bKcpNgLRRAW87qugiSncmycKIPypHCh5nH
From: "1060"sip:1060@mydomain.tld;tag=U5d8C1sd4CRTkQQHTiCh
To: sip:14172583@mydomain.tld;tag=as23f5b0df
Call-ID: 2b9ea5b5-41e9-30ac-fbcc-3f83e9f2ef1e
CSeq: 2264 INVITE
Content-Length: 0
Via: SIP/2.0/TCP 10.0.2.2:49497;rport;branch=z9hG4bKcpNgLRRAW87qugiSncmycKIPypHCh5nH;ws-hacked=WS
Server: Asterisk PBX
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH,MESSAGE
Supported: replaces,timer

tsk_utils.js?svn=224:116 SEND: ACK sip:14172583@mydomain.tld SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKcpNgLRRAW87qugiSncmycKIPypHCh5nH;rport
From: "1060"sip:1060@mydomain.tld;tag=U5d8C1sd4CRTkQQHTiCh
To: sip:14172583@mydomain.tld;tag=as23f5b0df
Call-ID: 2b9ea5b5-41e9-30ac-fbcc-3f83e9f2ef1e
CSeq: 2264 ACK
Content-Length: 0
Route: sip:mydomain.tld:5060;lr;sipml5-outbound;transport=udp
Max-Forwards: 70

tsk_utils.js?svn=224:116 State machine: c0000_Outgoing_2_Terminated_X_i300_to_i699INVITE
tsk_utils.js?svn=224:116 === INVITE Dialog terminated ===
tsk_utils.js?svn=224:116 PeerConnection::stop()
tsk_utils.js?svn=224:116 ==session event = i_ao_request
tsk_utils.js?svn=224:116 ==session event = terminated
tsk_utils.js?svn=224:122 The FSM is in the final state

13

One option I have yet to configure in my sip.conf is the preferred_codec_only(?) setting as I am not very familiar with it. How would this affect the SDP offers? would it fully restrict the offer and prevent VP8 from being in the SDP offer?

14

It would have no effect at all on the offer made by Mozilla, as it is not part of the Mozilla configuration.

15

I’ve tested with chrome as well, the SDP is in my post above, and am getting the same result. the only video codec being negotiated is VP8. Has chrome also removed support for H26x codecs?

16

WebRTC use VP8 for video, Asterisk can make a passtrough but i think your Polycom device doesn’t handle it, if you want to transcode video then you will need a media gateway.

17

Hi Navaismo, I’ve used your guides as a starting point for the project i’m working on, thank you for the work you did.

I was able to resolve the issue actually, I am using a media server, webrtc2sip, but was hoping I could make calls without it since asterisk, the browser and the polycom units support the h26x codecs but they don’t seem to be able to speak the same language. When using webrtc2sip I had to adjust the order of preference for the video codecs until I found one that worked, it will sometimes choose VP8 and transcode it and at other times will choose H263 but it still requires transcoding for video to flow both ways. I’m not exactly sure what the difference is in the three implementations of H26x but at least it’s flowing.

Any tips on improving video quality? I’m running on a VM at the moment and I think hardware is a partial issue in poor quality with all the overhead transcoding takes. What specs should I be requesting for a server that will be running Asterisk, Webrtc2SIP and hosting up to 50 video clients?